Many operators of existing 2G cellular telecommunication networks have now introduced packet switched based data services. In GSM networks, these services are facilitated by the General Packet Radio Service (GPRS) protocols and systems. A typical network architecture is illustrated in FIG. 1. The 3G standards (produced by the 3rd Generation Partnership Project) have introduced the concept of a Media Resource Function (MRF) 1 which is intended to act as a general purpose media handling node and might typically be located within an IP Multimedia Sub-System (IMSS) 4 of the cellular network. MRF nodes may also be introduced into 2G networks offering packet switched based data services.
One specific function of the MRF is in the handling of Voice over IP (VoIP) conference calls including the mixing and distribution of user media. In an example architecture, participants 3 in a conference call establish the call using Session Initiation Protocol (SIP) signalling which is facilitated and routed through the IMSS 4. A SIP server 5 known as the Serving Call Session Control Function (S-CSCF) routes SIP signalling to and from the MRF 1 in order to establish and control calls. Once a session has been established, media is routed between the MRF and user terminals (referred to below as “User Equipment” or UEs) via the Radio Access Network (RAN) 6 and the GPRS core network 7 (in particular via the GPRS Gateway Support Nodes (GGSNs) 8). NB. In FIG. 1 only connections between a first of the UEs 3 and the network elements are shown in detail. The connections between the network and the other UEs (identified by dotted lines) merely indicate the exchange of data (i.e. respective RANs, IMSS, etc are omitted).
The Real time Transport Protocol (RTP) is an Internet protocol standard that defines a way for applications to manage the real-time transmission of multimedia data. RTP is used at the bearer or media level (as opposed to the call control level which employs SIP or other call control protocol) for Internet telephony applications including VoIP. RTP does not guarantee real-time delivery of multimedia data, as this is dependent on the actual network characteristics. RTP provides the functionality to manage the data as it arrives to best effect. User Plane Adaptation (UPA) is the procedure used by the MRF and a given UE to monitor the RTP traffic between them and to adjust bandwidth utilisation in an attempt to provide optimal quality during a talk session. UPA provides for the MRF to dynamically redefine the talk burst duration which is encapsulated in a given RTP packet on a given link (this parameter is known as ptime) and the codec used for that link (the codec is identified by one of a number of parameters contained in a “mode set”). The SIP message reINVITE/UPDATE is used to signal these parameters to the UE. The UE may also send this message to the MRF in order to notify the MRF of its capabilities/requirements.                The group known as the Open Mobile Alliance has developed a Push to talk Over Cellular (PoC) specification aimed at enabling the provision of services over standard mobile networks which resemble walkie-talkie services, i.e. at the push of a button a subscriber can be instantly connected to one or more other subscribers. PoC relies upon the MRF to set up and handle connections. The PoC specification describes the tools available to detect packet loss over the links between the MRF and individual UEs. PoC also describes a means to request a change in bandwidth utilization, but does not provide detailed algorithms or procedures to enable this.        